In a digital audio system for processing incoming digital audio signals generated by a variety of independent sources, such as by compact disc (CD) players and digital audio tape (DAT) players/recorders, it is known to process these independently generated digital audio signals by a centralized digital signal processor (DSP).
The central DSP processes the incoming digital information from the one or more digital audio sources. This processing typically includes functions such as audio volume control, balance, muting, graphic equalization, and the like, as well as more sophisticated audio effects such as surround-sound decoding, concert hall simulation, mixing, and reverberation. The processed digital information is then converted into low power analog signals, which are subsequently amplified by a high power amplifier for application to speakers.
A standardized format generally used in the digital audio industry for digitally encoding analog audio signals and transmitting these signals in a serial format is the Audio Engineering Society (AES) / European Broadcasting Union (EBU) standard. In the AES/EBU standard, each sample of a stereo (two channel) audio signal is encoded into a frame containing 64 bits. A typical sampling rate is 44.1 kHz. Thus, 44,100 frames are generated each second. Each frame consists of two subframes 32 bits wide, wherein a first subframe contains up to a 24 bit linearly represented sample of a first channel of audio information, and a second subframe contains up to a 24 bit linearly represented sample of a second channel of audio information.
The AES/EBU standard provides space in each subframe for conveying additional information, including a preamble, a validity bit, a user data bit, a channel status bit, and a parity bit. The organization of a frame format and the function of these additional information bits are described in the article entitled "AES Recommended Practice for Digital Audio Engineering--Serial Transmission Format for Linearly Represented Digital Audio Data," Journal of the Audio Engineering Society, Vol. 33, No. 12, December 1985. This article is incorporated herein by reference.
Typically, the digital audio signals supplied to the central DSP from the various independent digital audio sources are each processed by the DSP in accordance with a different sampling rate clock signal, since the various digital audio signals are not synchronized with one another. Thus, in the prior art, a number of separate clock signals must be used to clock the circuitry within the DSP for concurrently processing incoming data signals generated by multiple independent digital audio sources. This type of prior art device results in inefficient and often complicated processing requirements and circuitry within the DSP.
What would be desirable is a method and structure to more efficiently process digital audio signals concurrently generated by two or more independent digital audio signal sources.